similar to: Asterisk switching bridge to native_rtp even with direct_media=no

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com> wrote: > Hey guys, > > have issues with reinvite, no matter what endpoint is calling asterisk > always tries switch simple_bridge to native_rtp > > Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge > technology to native_rtp > > in endpoints table ?direct_media? sets to ?no? on

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: > Well, it breaks audio for all NAT endpoints, how can I fix this? > Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew Jordan Digium, Inc. | Director of Technology

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

I have a hosted server. I have TWO different locations what have phones. Chicago and Indiana If I send audio direct from server to Chicago I hear it - same with indiana. But if indiana calls chicago - NO AUDIO. I see this in the CLI -- Channel SIP/63009-00000013 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> -- Channel SIP/63000-00000012 joined

On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <cursor at telecomab.mx> wrote: > Some users have complained that their calls drop after about 30 > seconds. Not all, just some. After looking at the log files the only > difference I can find from the dropped calls is the following line: > > [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge >

Hi friends, I am just starting use asterisk for our VoIP server. It works fine in LAN. But when it is deployed in public network(with a public IP), the SIP clients in different NAT fails to communicate with each other. I have set 'icesupport' to 'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails! Hope someone to help me out! Thanks in advance:) This

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP